HT801 - Grandstream Singapore
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HT801

1-Port ATA

The HT801 is a compact single port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution.

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Description

The HT801 is a single port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection and auto provisioning options enable users to take advantage of VoIP on analog phones. It also allows service providers to offer high quality IP service to their market. The HT801 is an ideal ATA for individual use as well as commercial IP voice deployments worldwide.

Features

  • Supports 1 SIP profile through a single FXS port and a single 10/100Mbps port
  • TLS and SRTP security encryption technology to protect calls and accounts
  • Automated provisioning options include TR-069 and XML config files
  • Supports 3-way voice conferencing
  • Failover SIP server automatically switches to secondary server if main server loses connection
  • Supports T.38 Fax for creating Fax-over-IP
  • Supports a wide range of caller ID formats
  • Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
  • Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multilanguage prompts, flexible dial plan and more
1 line
1 line
Caller ID
Caller ID
ENCRYPTION
Encryption security
Auto provisioning
Auto provisioning
3 way conference
3 way conference
Failover
Failover
Fax support
Fax support

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Designed for users looking to connect their analog devices to a VoIP network, in either a home or office. The HT801 is a powerful analog telephone adapter that is easily deployable and manageable. It comes equipped with 1 FXS ports to create a high-quality network solution.

Features

  • Supports 1 SIP profile through a single FXS port and a single 10/100Mbps port
  • TLS and SRTP security encryption technology to protect calls and accounts
  • Automated provisioning options include TR-069 and XML config files
  • Supports 3-way voice conferencing
  • Failover SIP server automatically switches to secondary server if main server loses connection
  • Supports T.38 Fax for creating Fax-over-IP
  • Supports a wide range of caller ID formats
  • Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
  • Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multilanguage prompts, flexible dial plan and more

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